said by brg:
Question 1: So, if I have a home asterisk implementation, and all my extensions are local to my personal private network, I wouldn't ever need to open any SIP port (say, 5060) on my router, nor would I need to forward said port to my asterisk server box and/or ATAs? That makes sense for local outbound calls.
Question 2: Same as above; but assume inbound calling to one of my DIDs from a VoIP provider. I'm registered to that provider from my asterisk box. No opening/forwarding of SIP port(s) to the * box needed? Because of my registration?
No forwarding for SIP ports, because of your registration AND keepalives.
You will need to forward your RTP ports if you want to forward incoming calls through your server.
Question 3: Assume inbound calling to one of my DIDs from a VoIP provider and I'm registered to that provider direct from my ATA -- no asterisk box. No opening/forwarding of SIP port(s) to the ATA needed? Because of my registration?
Correct, because of your registration AND keepalives.
Question 4: Now I'm traveling and want to connect via SIP client on my iPod to my asterisk; have that client register as an authorized extension. Is this the only situation requiring opening/forwarding of SIP port(s) to the * box? (Yes, I'm aware of "the traveling man", etc...)
Yes, you will have to open a port.